The present invention relates to an apparatus and method for transmitting data blocks based on priority from a transmitter in a UMTS (Universal Mobile Telecommunications System) type IMT-2000 system, and in particular, to an apparatus and method for transmission whereby a particular protocol layer determines the priority of each data block among the data blocks received from an upper layer through a single data stream, transfers the determined priorities together with the data blocks to a lower layer through a single data stream, and the lower layer receiving the data blocks with their priorities guarantees the respective quality of service (QoS) according to each respective priority for each data block for transmission over the radio interface, to thus guarantee a respectively different QoS for each data block within a single data stream having the same QoS being guaranteed.
A universal mobile telecommunication system (UMTS) is a third generation mobile communications system that has evolved from the European Global System for Mobile communications (GSM) that aims to provide an improved mobile communications service based upon a GSM core network and wideband code division multiple access (W-CDMA) wireless connection technology.
FIG. 1 illustrates an exemplary basic architecture of a UMTS network. As shown in FIG. 1, the UMTS is roughly divided into a terminal 100 (mobile station, user equipment (UE), etc.), a UMTS Terrestrial Radio Access Network (UTRAN) 120, and a core network (CN) 130. The UTRAN 120 includes one or more radio network sub-systems (RNS) 125. Each RNS 125 includes a radio network controller (RNC) 123, and a plurality of base stations (Node-Bs) 121 managed by the RNC 123. One or more cells exist for each Node B 121.
The RNC 123 handles the assigning and managing of radio resources, and operates as an access point with respect to the core network 130. The Node-Bs 121 receive information sent by the physical layer of the terminal 100 through an uplink, and transmit data to the terminal through a downlink. The Node-Bs 121, thus, operate as access points of the UTRAN 120 for the terminal 100. Also, the RNC 123 allocates and manages radio resources and operates as an access point with the core network 130. Between various network structure elements, there exists an interface that allows data to be exchanged for communication therebetween.
FIG. 2 illustrates a radio interface protocol architecture that exists in the mobile terminal and in the UTRAN as one pair, for handling data transmissions via the radio interface. Regarding each radio protocol layer, the first layer (Layer 1) is a physical layer (PHY) that serves the purpose of transmitting data over the radio interface by using various radio transmission techniques. The PHY layer is connected with an upper layer, the MAC layer via transport channels, which include a dedicated transport channel and a common transport channel depending upon whether that channel is shared or not.
In the second layer (Layer 2), a medium access control (MAC) layer, a radio link control (RLC) layer, a packet data convergence protocol (PDCP) layer and a broadcast/multicast control (BMC) layer exist. The MAC layer serves the purpose of mapping various logical channels to various transport channels, as well as performing logical channel multiplexing for mapping a plurality of logical channels to a single transport channel. The MAC layer is connected to a higher layer (e.g., the RLC layer) via logical channels, and these logical channels are divided into control channels that transmit control plane information and traffic channels that transmit user plane information.
The RLC layer handles the guaranteeing of the quality of service (QoS) of each radio bearer (RB) and the transmission of the corresponding data thereof. To guarantee the unique QoS of a radio bearer, the RLC layer has therein one or two independent RLC entities for each radio bearer, and provides three types of RLC modes; a transparent mode (TM), an unacknowledged mode (UM), and an acknowledged mode (AM), in order to support the various QoS. Also, the RLC layer adjusts the data size accordingly such that a lower layer may transmit data over the radio interface, by performing segmentation and concatenation on the data received from an upper layer.
The PDCP layer is located above the RLC layer and allows data that is transmitted by using Internet Protocol (IP) packets, such as IPv4 or IPv6, to be effectively transmitted over a radio interface having a relatively smaller bandwidth. For this purpose, the PDCP layer performs a header compression function, whereby only the absolutely necessary data in the header portion of the data are transmitted, in order to increase transmission efficiency over the radio interface. Because header compression is its basic function, the PDCP layer only exists in the PS (packet switched) domain, and a single PDCP entity exists per each radio bearer (RB) for providing effective header compression function with respect to each PS service.
Additionally, in the second layer (L2), a BMC (Broadcast/Multicast Control) layer exists above the RLC layer for performing the functions of scheduling cell broadcast messages and broadcasting to terminals located in a particular cell.
The radio resource control (RRC) layer located at the lowest portion of the third layer (L3) is only defined in the control plane, for controlling the parameters of the first and second layers and for controlling the transport channels and the physical channels in relation to the establishment, the re-configuration, and the releasing of the radio bearers (RBs). Here, the RB refers to a logical path provided by the first and second layers of the radio protocol for data transfer between the terminal and the UTRAN. And in general, the establishment of a radio bearer (RB) refers to regulating the protocol layers and the channel characteristics of the channels required for providing a specific service, as well as setting their respective specific parameters and operation methods.
Hereafter, the establishment of a radio bearer according to a quality of service (QoS) will be explained. QoS refers to the quality of service that the end-user notices upon being provided with a particular service. Various factors affect the QoS, such as delay time, error ratio, bit rate, and the like. In UMTS, an appropriate QoS is determined according to the type of service that is to be provided to the end-user. Here, the appropriate QoS refers to a minimum QoS that allows the end-user to be provided with the service, and the reason for setting a minimum QoS is to allow the service to be provided to a plurality of users. Namely, because radio resources are limited, providing a service using a high QoS to a particular user means that a large amount of radio resources are allocated to that particular user, and thus the total number of users to which service can be provided by the UMTS is decreased when considering the overall cell in which service is being provided.
In UMTS, as the entity that determines the QoS for a certain type of service, a MSC (Mobile Switching Center) is used for CS (circuit switched) services, while a SGSN (Serving GPRS Supporting Node) or a GGSN (Gateway GPRS Supporting Node) is used for PS (packet switched) services, and these entities exist within the core network (CN). When the QoS determining entities receive a request for a particular service from a terminal or an entity outside of the UMTS, an overall QoS is determined between the terminal and the QoS determining entity.
FIG. 3 depicts how the QoS between the terminal (UE) the Node B/RNC and the MSC (SGSN/GGSN) are defined. In FIG. 3, the QoS is separately established by sections, which can be broadly divided into a “lu section” that is a wired (wireline) interface between the MSC (or the SGSN/GGSN) to the Node B/RNC, and a “Uu section” that is a wireless (radio) interface between the Node B/RNC and the terminal. Also, the lu section has a lu Bearer and the Uu section has a radio bearer (RB) established, respectively, for providing services having an appropriate QoS. The overall QoS between the terminal and the MSC (or the SGSN/GGSN) is determined by the sum of the quality of service for the lu interface (“QoS-lu”) and the quality of service for the Uu interface (“QoS-Uu”). As the wireless interface has a more disadvantageous environment when compared to that of the wired interface, the overall QoS mostly depends upon the QoS-Uu.
The details of the QoS and bearer configuration procedures will be explained with respect to a VoIP (Voice over Internet Protocol) service as being a representative example of a PS service. First, assuming that the SGSN received a request for VoIP service from a terminal, the SGSN determines an appropriate QoS for providing the requested VoIP service by considering the priority and/or capabilities of the terminal and/or considering various types of available resources. Also, it is assumed that the SGSN determined the overall QoS with the following parameters: Delay=200 ms; Error Ratio=10−2; Bit Rate=36 kbps. Based upon this overall QoS, the SGSN then determines the QoS for each section. Here, because the wired interface generally has a more advantageous environment compared to that of the wireless interface, it does not greatly affect the overall QoS. Namely, the wired interface has a delay of less than a few milliseconds (ms), an error ratio of less than 10−6 and a bit rate of several to several hundred megabits per second (Mbps), thus most of the values for the overall QoS are directly applicable to the QoS-Uu. Generally, the error rate and bit rate of the overall QoS are directly applied to the QoS-Uu, and a delay value that excludes a few milliseconds (ms) needed for the core network protocol to process data is applied. Thus, for this situation, it is assumed that the SGSN determined the QoS-Uu to have the following parameters: Delay=180 ms; Error Ratio=10−2; Bit Rate=36 kbps. Then, the SGSN informs this determined QoS-Uu to the RNC, and the RNC configures an appropriate radio bearer (RB) in accordance thereto.
The RNC configures the RB based upon the QoS-Uu informed by the SGSN. More accurately, the RRC layer, which is a radio protocol layer within the RNC, configures the RB. As explained previously, the RB refers to a logical path provided by the first and second layers of the radio protocol, and the data transmitted through the RB are basically guaranteed the quality corresponding to the QoS-Uu. In order to configure an RB that satisfies the QoS-Uu, the RRC layer of the RNC configures the first and second layers of the radio protocol, and various characteristics, operation procedures, parameters, and the like for each of the channels. For example, with respect to the PDCP layer, the type of header compression method to be used, etc. are determined. With respect to the RLC layer, the type of operation mode to be used, the maximum data storage time to be used, the size of the RLC PDU (protocol data unit) to be used, various timer values and protocol parameter values to be used, etc. are determined. With respect to the MAC layer, the type of channel mapping to be used, the method of channel multiplexing to be used, the method of priority processing to be used, how to perform transmission format combinations, etc. are determined. With respect to the PHY (physical) layer, the method of modulation to be used, the coding methods to be used, the type of CRC (cyclic redundancy check) to be used, the transmission power level to be used, the types of physical channels to be used, etc. are determined.
After the RRC of the RNC determines all of the aspects of the RB, the first and second layers of the RNC are established according to the determined aspects, and simultaneously informs these aspects to the RRC of the terminal to allow the first and second layers of the terminal to be established according to these aspects. When the RB is established in this manner, a logical path between the terminal and the RNC is formed, and thereafter, data is transmitted according to the determined path. Here, as explained before, because a single RB guarantees a single QoS-Uu, the data transmitted through the same RB are all guaranteed the same QoS-Uu.
In the related art, because a single RB guarantees a single QoS (QoS-Uu), the data transmitted through the same RB is guaranteed the same QoS. However, there are certain situations where the data transmitted via a single RB will have respectively different priorities according to the processing techniques of the radio protocol layers, thus requiring respectively different QoS to be guaranteed. The header compression performed at the PDCP layer is an example of one such situation.
The header compression technique utilizes the fact that many portions of the IP headers of IP packets that are part of the same packet stream do not change at all or do not change very often. The fields that do not change are stored in the format of context within a compressor of the transmitting side (i.e., transmitter) and within the decompressor of the receiving side (i.e., receiver), and the overhead of an IP header can be reduced by only transmitting those fields that have changed after the context has been formed. During the initial stages of header compression, because the compressor transmits full header packets to the decompressor in order to form the context with respect to the corresponding packet stream, there is no gain (advantage) of using header compression. But after the context is formed in the decompressor, the compressor only transmits compressed header packets, and thus the gain (advantage) is drastic.
For a particular packet stream, determining whether to transmit a certain packet with a full header or with a compressed header can be entirely dependent upon the compressor. However, in general, when context is to be initially formed for a particular packet stream, one or more full header packets should be transmitted. As compressed header packets are transmitted thereafter, upon the lapse of a certain time period, one or more full header packets are transmitted intermittently such that the context of the decompressor is maintained in synchronization with the context of the compressor.
FIG. 4 depicts an example of how full header packets and compressed header packets are transmitted when using a header compression technique. When the compressor of the PDCP in the transmitter receives an IP packet from an upper layer, the corresponding packet is transmitted to the receiver as a full header packet or a compressed header packet according to the pattern of the header. If it is determined that there is a need to form a new context or a need to update the context, the compressor transmits the packet as a full header packet. If it is determined that the context with respect to the header pattern of the corresponding packet is already formed in the decompressor, then the compressor transmits the packet as a compressed header packet.
The decompressor of PDCP in the receiver forms a context by first receiving a full header packet for a certain packet stream. This is because the context will be the basis from which the compressed headers to be received will be decompressed. If the decompressor receives compressed header packets in a state where the context has not been formed, the decompressor cannot decompress the original header of the corresponding packet and thus will discard that received packet.
As such, when a header compression technique is used in a radio interface for a certain PS service, the PDCP in the transmitter transmits the IP packets that were received from an upper layer in a single stream having the same QoS, in either a “packet for forming or updating context” format or a “a packet for not forming or updating context” format. However, if a “packet for forming or updating context” is not successfully transmitted to the receiver, all subsequently transmitted “packets for forming or updating context” cannot be decompressed at the receiver and are thus discarded. Thus, a “packet for forming or updating context” is relatively much more important (i.e., has higher priority) than a “packet for not forming or updating context”.
However, in the related art, all the data transmitted via a single RB has the same QoS, and relatively more important data cannot be transmitted with a higher QoS when compared to relatively less important data. Thus, there is a need for allowing data to be transmitted with different QoS according to its importance (i.e., priority), even though the data is transmitted via a single RB.
Thus, the inventors of the present invention recognized such drawbacks of the related art and provided a solution by providing a particular protocol layer of the transmitting side (transmitter) that initially receives service data units (SDUs) having the same priority through a single stream from an upper layer, processes these SDUs to generate protocol data units (PDUs) having different priorities, and uses respectively different transmission methods to transmit the generated PDUs over a radio interface in order to guarantee their respectively different quality of service (QoS) requirements.